1. Field of the Invention
The present invention relates to a method of controlling a variable bit-rate codec, and more particularly, to a method of controlling a variable bit-rate codec capable of securing QoS of a voice service between terminals by controlling a transmission rate of the variable bit-rate codec based on transmission capability of a network, when a real time multimedia service is provided through a linkage between a packet network and an existing wired and wireless network and when using a variable bit-rate codec that provides different transmission rates to transmitting and receiving ends.
The present invention was supported by the Information Technology (IT) Research & Development (R&D) program of the Ministry of Information and Communication (MIC) [project management number: 2005-S-100-02, subject title: Development of Multi-codec and Its Control Technology Providing Variable Bandwidth Scalability].
2. Description of the Related Art
A variable bit-rate codec is a technique of converting a natural sound into digitally transformed codec data with a plurality of transmission rates. For example, a frequency band can be classified into a narrow-band (a band ranging form 300 Hz to 3400 Hz), a wide-band (a band ranging from 50 Hz to 7000 Hz), or an audio-band (a band ranging from 20 Hz to 20000 Hz). In each band, transmission rates of 8, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30, and 32 kbps can be obtained.
For example, a bandwidth provided by a network of a voice over internet protocol (VoIP) voice telephone service in a packet network is variable and unpredictable. In addition, in variable bit-rate codec with a transmission rate of 32 kbps, high-quality sounds are produced. In variable bit-rate codec with a transmission rate of 8 kbps, low-quality sounds are produced. In this case, when there is a spare band in the network and when it is possible to transmit a signal within a high frequency band, a signal with a transmission rate of 32 kbps can be transmitted. When the network band is changed and a network situation deteriorates, a signal with a low transmission rate such as 30 kbps, 28 kbps, or the like can be transmitted. Although the sound quality of signals with lower transmission rates is deteriorated, it is possible to achieve good transmission in the network.
In a variable bit-rate codec, when a transmission rate is high, sounds have high quality, but the probability of loss and delays in transmission through the network is high. On the contrary, when the transmission rate is low, sounds have low quality, but the probability of loss and delays in transmission through the network is low.
On the other hand, in order to apply a variable bit-rate codec, a signal protocol conversion technique for making a call is applied. A signal protocol conversion technique is disclosed in RFC 3261 SIP of IETF, RFC 3264 Offer/Answer SDP, RFC 2833 RTP Payload for DTMP Digits, Telephony Tones and Telephony Signals, RFC 2327 SDP, RFC 3108 ATM SDP, RFC 1890 RTP Profile Payload type, and the like.
An apparatus for calling by selecting a codec determined by network parameters among a plurality of codecs installed in a terminal is disclosed in U.S. Pat. No. 7,002,992 B1 (“Codec Selection to improve media communication”).
An adaptive multirate (AMR) codec control method of controlling a bit-rate of a voice codec based on strength of a wireless signal is disclosed in U.S. Pat. No. 2003/0189900 A1 (“Communications using adaptive multi-rate codecs”).
However, the aforementioned conventional technique does not control the bit-rate of the codec based on voice quality.